My guide (SACD Ripping) walks you through the settings I recommend in Korg Audiogate. The sample rate is up to you, based on the sweetspot of your DAC, etc. but basically when I used to convert DSD I used soft rolloff, Aqua dithering and I also used "normalize songlist" so Audiogate would make sure not to clip certain hot DSD titles.

There is some code out there for Linux fans to use (DSD2PCM I think it's called). However, many folks simply let the player do it on the fly (thereby not having to permanently store additional PCM files). Players like A+, PM, JRiver and HQplayer all use very good DSD-to-PCM computations to do the conversion. This, of course, is a perfect solution if you someday plan to do a DSD-capable DAC.

mlknez, have you compared foobar2000 with the SACD plugin to Korg Audiogate? I've using Audiogate and the results are great, but would like to compare it to foobar2000.

I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.

My guide (SACD Ripping) walks you through the settings I recommend in Korg Audiogate. The sample rate is up to you, based on the sweetspot of your DAC, etc. but basically when I used to convert DSD I used soft rolloff, Aqua dithering and I also used "normalize songlist" so Audiogate would make sure not to clip certain hot DSD titles.

There is some code out there for Linux fans to use (DSD2PCM I think it's called). However, many folks simply let the player do it on the fly (thereby not having to permanently store additional PCM files). Players like A+, PM, JRiver and HQplayer all use very good DSD-to-PCM computations to do the conversion. This, of course, is a perfect solution if you someday plan to do a DSD-capable DAC.

Dear Ted,
thx for your answer. I have read your guide and saw your recommencation. But wanted some other opinions. At the moment I have foobar2000 + sacd-plugin.
JRMC is not an option for me, my Atom330 has not enough power for realtime conversation.
Did you (or anybody else) ever compare these two? Or is there a third option? - not the expensive saracon.

mlknez, have you compared foobar2000 with the SACD plugin to Korg Audiogate? I've using Audiogate and the results are great, but would like to compare it to foobar2000.

I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.

Gary

Hi Gary,

I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

Bernhard

From some (very good) recording engineer I learned: The 0db limit on PCM recordings it's not so important as in DSD recordings. Of course this don't mean to get clipping all the time in PCM.

I don't really see the point in converting to PCM unless you need them for portable playback.

Converting to PCM is a potentially lossy process, so you are best to use a player that can handle realtime DSD to PCM playback without altering the source files.

JRiver Media Center seems to have some good options for this, though I used a custom 30kHz low-pass filter with a 48dB/octave slope instead of the 24dB/octave slope preset, as this filters out all the ultrasonic noise.

Now that I have a DSD-capable DAC, I just bitstream the DSD files to it. (this is why I keep the source)

Skeptic,
This thread is for the majority of CA folks, those without a DSD-capable DAC. Of course converting to PCM for no reason...is not recommended.

Up until a few weeks ago, I was without a DSD-capable DAC too - but that's why I recommend leaving it as native DSD and having the player convert to PCM in realtime, rather than destructively converting the file to PCM.

Then if you ever do buy a device that supports native DSD, you can take advantage of it.

That said, I do think the recent DSD hype is being overblown, and I wouldn't buy new hardware just to get native DSD support.

I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.

I set the gain to the default 0 dB, load the ripped .iso, then run the DRM plug-in on it (it takes a while), then use the resulting peak numbers to set the gain accordingly before running the conversion. Note that you have to clear out the .iso and reload it for the new setting to take effect.

I've come across a case where there is some weird clipping going on, apparently with DSD->PCM conversion. The track in question is track 2 of the new Audio Fidelity America - America release... Sandman. The left waveform is and SACD rip converted to PCM. It looks the same with AudioGate or the foobar plugin. The right waveform, is a rip of the CD layer.

As one can see the peak is clipped off on the DSD->PCM conversion. I tried various gain settings in AudioGate with no change. Anyone have any theories as to what is going on here?

I recorded the analog output of both the SACD and CD from an Oppo BDP-93 and didn't see any differences there so it appears to be something going on in DSD->PCM conversion.

There's some disagreement about DSD vs PCM levels. 50% modulation is the traditional peak level for DSD, but the SACD specification allows exceeding this. It's also possible the DSD encoding baked the clip in (range handling to avoid overflow of the noise shaper).

JRiver supports up to 100% modulation from a DSD file before it would clip in the integer domain. And since JRiver uses all 64-bit floating point math, you can attenuate to avoid clips by using R128 Volume Leveling (including dealing with intersample overs).

You might try JRiver and use the 'Analyze Audio' tool to determine the peak level.

Some well known DAC chips clip on the analog output after 50% modulation when playing DSD.

I would guess these are the ones that do digital processing for DSD... Some DAC chips also clip on inter-sample overs too...

And some DSD files in the wild reach almost 100% modulation.

Those shouldn't be coming from SACD masters at least, because SACD pressing plants should refuse masters exceeding the +3.1 dB level. For example HQPlayer keeps watch and prevents output exceeding the spec level. Maybe some modulators are not keeping eye on the output.

Not that it would be problem for HQPlayer's PCM conversion either, because there's practically no max sample value for the processing pipeline.